July 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Add several entries for enabling HP mute led
by Hui Wang 07 Jul '14

07 Jul '14

BugLink: https://bugs.launchpad.net/bugs/1329580 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- So far, the mute led DMI string on these machines are HP_Mute_LED_0_0. sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 832c225..327413e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4854,6 +4854,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x21f8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -4907,6 +4909,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8004, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), /* ALC290 */ SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221c, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221d, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2220, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2222, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Use devm_snd_soc_register_component
by Manish Badarkhe 07 Jul '14

07 Jul '14

Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe <badarkhe.manish(a)gmail.com> --- :100644 100644 efe2cd6... bfc8f8e... M sound/soc/omap/omap-mcbsp.c sound/soc/omap/omap-mcbsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index efe2cd6..bfc8f8e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -805,7 +805,7 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); if (ret) return ret; @@ -817,8 +817,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); - if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); -- 1.7.10.4

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[alsa-devel] [PATCH v5 0/5] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 07 Jul '14

07 Jul '14

Hi Lars, Mark and Liam, Here is the fifth version of the multicodec series. It mostly fixes the remaining comments from Lars on the series. It removes as well the support for soc-compress.c based on Vinod comment. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test it on various platforms to ensure this is not generating any regression. The series is based on asoc/for-next (v3.16-rc3). Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: http://comments.gmane.org/gmane.linux.alsa.devel/121024 - Split the first version in several patches. - Remove the TDM fixup v3: - Fixed all the places that were not iterating over all the codecs. - Change the structure and split it. v4: - Improve the TDM management - Split the orginal patch in smaller chunks - Ignore the CODEC to CODEC link case for the moment - Add iteration for multicodecs everywhere v5: - Remove multicodec support for soc-compress - Simplify the code a little thanks to Lars component series Benoit Cousson (5): ASoC: core: Add initial support for DAI multicodec ASoC: pcm: Add support for DAI multicodec ASoC: dapm: Add support for DAI multicodec ASoC: compress: Prevent multicodec for compressed stream ASoC: pcm: Add soc_dai_hw_params helper include/sound/soc-dai.h | 4 + include/sound/soc.h | 17 ++ sound/soc/soc-compress.c | 5 + sound/soc/soc-core.c | 293 +++++++++++++++++-------- sound/soc/soc-dapm.c | 95 ++++---- sound/soc/soc-pcm.c | 555 ++++++++++++++++++++++++++++++++--------------- 6 files changed, 665 insertions(+), 304 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 00/19]: add mrfld DSP support
by Vinod Koul 07 Jul '14

07 Jul '14

Here is the split patch series for adding DSP support for Intel's merrfield platform. The last patch si mxer update patch which we have been discussing over the list. This is only for discussion and sake of complteness. For that patch we need the dapm_kcontrol_get/set series to do merged OR patch to be reworked after compenent series Vinod Koul (19): ASoC: Intel: add COMPILE_TEST to mfld machine ASoC: Intel: mfld_pcm: move stream handling to dai_ops ASoC: Intel: mfld-pcm rename period callback arg ASoc: Intel: mfld-pcm: report pcm delay ASoC: Intel: add the mrfld fw IPC definations ASoC: Intel: mfld-pcm: modularize stream allocation code ASoC: Intel: add mrfld pipelines ASoC: Intel: use common stream allocation method for compressed stream ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: add mrfld DSP registers ASoC: intel: mfld-pcm: don't call trigger ops to DSP for internal streams ASoC: Intel: add generic parameter set interface ASoC: Intel: mrfld: add bytes control for modules ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mfld-pcm: add control for powering up/down dsp ASoC: Intel: mrfld: add DSP core controls ASoC: Intel; mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add the fe & be dai ops ASoC: Intel: mrfld: add the DSP mixers arch/x86/include/asm/platform_sst_audio.h | 78 ++ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/Makefile | 3 +- sound/soc/intel/sst-atom-controls.c | 1386 ++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 906 +++++++++++++++++ sound/soc/intel/sst-mfld-dsp.h | 414 ++++++++- sound/soc/intel/sst-mfld-platform-compress.c | 11 +- sound/soc/intel/sst-mfld-platform-pcm.c | 492 +++++++--- sound/soc/intel/sst-mfld-platform.h | 47 +- 9 files changed, 3199 insertions(+), 140 deletions(-) create mode 100644 arch/x86/include/asm/platform_sst_audio.h create mode 100644 sound/soc/intel/sst-atom-controls.c create mode 100644 sound/soc/intel/sst-atom-controls.h

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[alsa-devel] [PATCH] ALSA: hda - Fix and neaten print_nid_path/debug_badness
by Joe Perches 07 Jul '14

07 Jul '14

print_nid_path has a possible buffer overflow if struct nid_path.path values are > 256. Avoid this and neaten the output to remove the leading ':' Neaten debug_badness to always verify arguments. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/pci/hda/hda_generic.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 589e47c..b293583 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -350,16 +350,16 @@ static void print_nid_path(struct hda_codec *codec, const char *pfx, struct nid_path *path) { char buf[40]; + char *pos = buf; int i; + *pos = 0; + for (i = 0; i < path->depth; i++) + pos += scnprintf(pos, sizeof(buf) - (pos - buf), "%s%02x", + pos != buf ? ":" : "", + path->path[i]); - buf[0] = 0; - for (i = 0; i < path->depth; i++) { - char tmp[4]; - sprintf(tmp, ":%02x", path->path[i]); - strlcat(buf, tmp, sizeof(buf)); - } - codec_dbg(codec, "%s path: depth=%d %s\n", pfx, path->depth, buf); + codec_dbg(codec, "%s path: depth=%d '%s'\n", pfx, path->depth, buf); } /* called recursively */ @@ -1700,9 +1700,11 @@ static int fill_and_eval_dacs(struct hda_codec *codec, #define DEBUG_BADNESS #ifdef DEBUG_BADNESS -#define debug_badness(fmt, args...) codec_dbg(codec, fmt, ##args) +#define debug_badness(fmt, ...) \ + codec_dbg(codec, fmt, ##__VA_ARGS__) #else -#define debug_badness(...) +#define debug_badness(fmt, ...) \ + do { if (0) codec_dbg(codec, fmt, ##__VA_ARGS__); } while (0) #endif #ifdef DEBUG_BADNESS

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[alsa-devel] USB-Audio.conf comment regarding asound.conf
by Anssi Hannula 06 Jul '14

06 Jul '14

Hi, USB-Audio.conf contains: > # DO NO EDIT; this is an internal ALSA file. > # If you want to add your own definitions, put them into /etc/asound.conf or > # ~/.asoundrc, with "cards." before the "USB-Audio", e.g.: > # > # cards.USB-Audio.pcm.use_dmix."NoiseBlaster 3000" no > # > # If your device requires such a definition to work correctly, please report it > # to <alsa-devel(a)alsa-project.org>. However, snd_config_hook_load_for_all_cards() will completely skip loading of /usr/share/alsa/cards/USB-Audio.conf if one has existing configuration elements in cards.USB-Audio. I was just hit with this while debugging an issue with a user. Should users just edit /usr/share/alsa/cards/USB-Audio.conf directly until such entries are added upstream, or should this be somehow fixed in alsa-lib to work as instructed? Related commits: http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=96da0c842d14b40c… http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=133819d420d962b6… -- Anssi Hannula

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[alsa-devel] [PATCH] amixer: Fix convert_prange1 not rounding correctly
by Josef Gajdusek 06 Jul '14

06 Jul '14

The convert_prange1 macro was not the exact inverse of convert_prange. This was causing the volume percentage set by 'amixer set' to differ from the one displayed by 'amixer get'. Signed-off-by: Josef Gajdusek <atx(a)atx.name> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index cf82892..577a2ca 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -201,7 +201,7 @@ static int convert_prange(long val, long min, long max) /* Function to convert from percentage to volume. val = percentage */ #define convert_prange1(val, min, max) \ - ceil((val) * ((max) - (min)) * 0.01 + (min)) + rint((val) * ((max) - (min)) * 0.01 + (min)) struct volume_ops { int (*get_range)(snd_mixer_elem_t *elem, long *min, long *max); -- 1.8.5.5

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[alsa-devel] [PATCH] ASoC: s6105-ipcam: Fix build for I2C dependencies
by Mark Brown 05 Jul '14

05 Jul '14

From: Mark Brown <broonie(a)linaro.org> The s6105-ipcam driver calls i2c_register_board_info() which means that not only does it need I2C to be enabled to be built but since that symbol is not exported it means that the driver also can't be linked as a module. Use a bool instead of a tristate and add a dependency on I2C to fix these issues. Commit b5dd60fd263 (ASoC: s6000: Allow to build when COMPILE_TEST is enabled) made this problem more visible since it enabled build of the driver on more common architectures but this has always been an issue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/s6000/Kconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig index 4823e1e4daef..f244a2566f20 100644 --- a/sound/soc/s6000/Kconfig +++ b/sound/soc/s6000/Kconfig @@ -15,8 +15,9 @@ config SND_S6000_SOC_I2S tristate config SND_S6000_SOC_S6IPCAM - tristate "SoC Audio support for Stretch 6105 IP Camera" - depends on SND_S6000_SOC + bool "SoC Audio support for Stretch 6105 IP Camera" + depends on SND_S6000_SOC=y + depends on I2C=y depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST select SND_S6000_SOC_I2S select SND_SOC_TLV320AIC3X -- 2.0.0

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[alsa-devel] [PATCH 1/2] ASoC: max98090: Add check for CODEC type
by Tushar Behera 04 Jul '14

04 Jul '14

CODEC type (MAX98090/MAX98091) can be specified from device-tree file, it can also be obtained from the CODEC during runtime. Add an explicit check to figure out if both are matching, else print a message warning about the same. Signed-off-by: Tushar Behera <tushar.b(a)samsung.com> --- sound/soc/codecs/max98090.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ca74f502..1ed06d9 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2215,6 +2215,7 @@ static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; + enum max98090_type devtype; int ret = 0; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2250,16 +2251,21 @@ static int max98090_probe(struct snd_soc_codec *codec) } if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { - max98090->devtype = MAX98091; + devtype = MAX98091; dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); } else { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); } + if (max98090->devtype != devtype) { + dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n"); + max98090->devtype = devtype; + } + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: samsung: Correct I2S DAI suspend/resume ops
by Sylwester Nawrocki 04 Jul '14

04 Jul '14

We should save/restore relevant I2S registers regardless of the dai->active flag, otherwise some settings are being lost after system suspend/resume cycle. E.g. I2S slave mode set only during dai initialization is not preserved and the device ends up in master mode after system resume. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa..5f9b255 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -920,11 +920,9 @@ static int i2s_suspend(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - } + i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); + i2s->suspend_i2scon = readl(i2s->addr + I2SCON); + i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); return 0; } @@ -933,11 +931,9 @@ static int i2s_resume(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); - } + writel(i2s->suspend_i2scon, i2s->addr + I2SCON); + writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); + writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } -- 1.7.9.5

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